Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: A method and system for analysis-by-synthesis encoding of an information signal is provide. The encoder (400) can include the steps of generating a first synthetic signal based on a first pitch-related codebook (402), generating a second synthetic signal based on a second pitch-related codebook (404), selecting a codebook configuration parameter based on the reference signal and the first and second synthetic signals, and conveying the codebook configuration for use in reconstructing an estimate of the input signal. The encoder can include an error expression having an error bias (506) and a prediction gain having a prediction gain bias (508) for determining the codebook configuration. The encoder can employ variable length coding and combinatorial subframe coding (600) for efficiently compressing the codebook configuration parameter and codebook related parameters for one or more subframes.

Abstract: A system (100) and method (200) of combining codewords is provided. The system can include a splitter (120) for splitting a first codeword (110) into a most significant bits part MSP (112) and a least significant bits part LSP (114), a combiner (130) for combining the MSP of the first codeword with a second codeword to produce a first group (132), and a concatenator (140) for concatenating the first group with the LSP to produce a second group (134), and multiplexing the first group with the second group to produce a multiplexed codeword (150). Bit-errors in the LSP correspond to decoding errors only in a codeword associated with the LSP, and not to decoding errors in other codewords.

Abstract: A method and apparatus for improving listener differentiation of talkers during a conference call is provided herein. Particularly, during a teleconference a node (101) will extend the bandwidth of received signals (e.g., speech). Each caller within the conference call will then have their voice projected by the node (101) to a particular spot in three-dimensional space.

Abstract: In a speech encoder/decoder a pitch delay contour endpoint modifier is employed to shift the endpoints of a pitch delay interpolation curve up or down. Parficularly, the endpoints of the pitch delay interpolation curve are shifted based on a variation and/or a standard deviation in pitch delay.

Abstract: A CELP encoder is provided that optimizes excitation vector-related parameters in a more efficient manner than the encoders of the prior art. In one embodiment, a CELP encoder optimizes excitation vector-related parameters based on a computed correlation matrix, which matrix is in turn based on a filtered first excitation vector. The encoder then evaluates error minimization criteria based on at least in part on a target signal, which target signal is based on an input signal, and the correlation matrix and generates a excitation vector-related index in response to the error minimization criteria. In another embodiment, a CELP encoder is provided that is capable of jointly optimizing and/or sequentially optimizing multiple excitation vector-related parameters by reference to a joint search weighting factor, thereby invoking an optimal error minimization process.

Abstract: A speech coder that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal. The speech coder generates a target vector based on an input signal. The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

Abstract: To address the need for choosing values of harmonic noise weighting (HNW) coefficient (?p) so that the amount of harmonic noise weighting can be optimized, a method and apparatus for performing harmonic noise weighting in digital speech coders is provided herein. During operation, received speech is analyzed to determine a pitch period. HNW coefficients are then chosen based on the pitch period, and a perceptual noise weighting filter (C(z)) is determined based on the harmonic-noise weighting (HNW) coefficients (?p).

Abstract: A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component.

Abstract: To address the need for choosing values of harmonic noise weighting (HNW) coefficient (?p) so that the amount of harmonic noise weighting cam be optimized, a method and apparatus for performing harmonic noise weighting in digital speech coders is provided herein. During operation, received speech is analyzed to determine a pitch period. HNW coefficients are then chosen based on the pitch period, and a perceptual noise weighting filter (C(z)) is determined based on the harmonic-noise weighting (HNW) coefficients (?p).

Abstract: A CELP encoder is provided that optimizes excitation vector-related parameters in a more efficient manner than the encoders of the prior art. In one embodiment, a CELP encoder optimizes excitation vector-related parameters based on a computed correlation matrix, which matrix is in turn based on a filtered first excitation vector. The encoder then evaluates error minimization criteria based on at least in part on a target signal, which target signal is based on an input signal, and the correlation matrix and generates a excitation vector-related index in response to the error minimization criteria. In another embodiment, a CELP encoder is provided that is capable of jointly optimizing and/or sequentially optimizing multiple excitation vector-related parameters by reference to a joint search weighting factor, thereby invoking an optimal error minimization process.

Abstract: A speech coder that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal. The speech coder generates a target vector based on an input signal. The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

Abstract: The invention provides a method of coding an information signal. An information signal is represented by a sequence of pulses. A plurality of pulse parameters are determined based on the sequence of pulses including a non-zero pulse parameter corresponding to a number of non-zero pulse positions in the sequence of pulses. The non-zero pulse parameter is coded using a variable-length codeword.

Abstract: The invention provides a method of coding an information signal. An information signal is represented by a sequence of pulses. A plurality of pulse parameters are determined based on the sequence of pulses including a non-zero pulse parameter corresponding to a number of non-zero pulse positions in the sequence of pulses. The non-zero pulse parameter is coded using a variable-length codeword.